[Serusers] SER + Asterisk

[Laurent BURGY]

Hi,
First of all, thanks a lot Alessio for your help ... I m not so far away 
from the end of my quest....
I m still encountering one problem.... When i pass call to the pstn from 
ser through asterisk, even if it succeeds, the value on my client is 
still 404... My mobile phone is ringing but i see on the Xlite screen 
404, i don't understand ...

thx
Laurent
Alessio Focardi wrote:

>Hello Laurent,
>
>Tuesday, April 20, 2004, 6:21:07 PM, you wrote:
>
>LB> I think i follow the instructions ... but should i "route"  ( so modify
>LB> the ser.cfg file)  some messages to my gateway to use the answering
>LB> machine or services provided by my gateway ...
>LB> And what about autocreatepeer=yes?
>
>surely later you will have to route in ser.cfg for outbound calls, checking first for
>user auth and grp assignement.
>
>Using the extention method you can direct sip addresses to services in
>asterisk like voicemail or voice prompts.
>
>that's my ser routing for outside calls
>
>
>
>                                        
>                                        record_route();                 
>                                               
>                
>                                        if (uri=~"sip:0[0-9]+@"){
>                                
>                                        if (!proxy_authorize("mydomain.com", "subscriber")) {proxy_challenge("mydomain.com", "0");sl_send_reply("403", "That's not your home");break;}; #fine proxy challenge
>                                        if (!is_user_in("credentials", "local")){sl_send_reply("403", "No permission for local calls");break;}; #fine invite                               
>                
>                                        rewritehostport("sip.mydomain.com:5090");
>                                        t_relay();
>                                        break;
>                                
>                                        }; #fine if uri sip:0 
>
>
>inside asterisk calls forwarded by ser are treated by this extention
>
>exten => _0.,1,Dial,Zap/g1/${EXTEN:1}|45|r
>exten => _0.,2,Congestion
>
>its important to notice that you have to block port 5090 for incoming
>ip requests ....
>
>Only ser will be allowed to talk to asterisk and forward calls, after
>user checking.
>
>Hope it helps
>
>LB> thx
>
>LB> Alessio Focardi wrote:
>
>  
>
>>>Hello Laurent,
>>>
>>>Tuesday, April 20, 2004, 4:45:41 PM, you wrote:
>>>
>>>LB> But my clients should register on SER or on Asterisk?
>>>
>>>On ser, then you will need to protect asterisk from unallowed pstn
>>>call, but that will come later on.
>>>
>>>
>>>LB> thx
>>>
>>>LB> Alessio Focardi wrote:
>>>
>>> 
>>>
>>>      
>>>
>>>>>Hello Laurent,
>>>>>
>>>>>Tuesday, April 20, 2004, 1:50:37 PM, you wrote:
>>>>>
>>>>>LB> In fact, my ser installation works fine...
>>>>>LB> I can pass call through asterisk in standalone...
>>>>>LB> The problem is to interconnect the 2, to register ( i don't know if it's
>>>>>LB> a right solution) the sipphones on SER and to go outside thanks to
>>>>>LB> asterisk...
>>>>>
>>>>>make asterisk use port 5090 for sip, then as a first step make
>>>>>asterisk register in ser as an extention.
>>>>>
>>>>>you can do this in asterisk's sip.conf
>>>>>
>>>>>example
>>>>>
>>>>>register => 10:password at sip.yourdomanin.com/10
>>>>>
>>>>>this tells asterisk to register extention 10 as address 109 at yourdomain.com
>>>>>
>>>>>dial 10 with a sip phone and you are in asterisk ... note that you
>>>>>should have an extention 10 defined, or it will not work.
>>>>>
>>>>>B> Thx
>>>>>
>>>>>LB> Alessio Focardi wrote:
>>>>>
>>>>>
>>>>>
>>>>>     
>>>>>
>>>>>          
>>>>>
>>>>>>>Hello Laurent,
>>>>>>>
>>>>>>>What you want to accomplish could be done, my advice is to setup a
>>>>>>>working installation of ser then you will continue with asterisk.
>>>>>>>
>>>>>>>At first make a simple installation of ser (no auth, no db maybe) and make
>>>>>>>your phones call each other.
>>>>>>>
>>>>>>>If you encounter specific problems and you want to have some help this
>>>>>>>is the right place.
>>>>>>>
>>>>>>>Good luck !
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>Tuesday, April 20, 2004, 1:29:41 PM, you wrote:
>>>>>>>
>>>>>>>LB> Hi,
>>>>>>>LB> I want to use SER as a sip Proxy and asterisk as a gateway to the PSTN
>>>>>>>LB> network ...
>>>>>>>LB> My sipphones are BudgetTone101 and i'm having trouble trying configure
>>>>>>>LB> them....
>>>>>>>LB> Indeed, i don't know if they should register on SER or not... I don't
>>>>>>>LB> know what kind of sip messages should be passed to my machine running
>>>>>>>LB> asterisk.
>>>>>>>LB>    I don't know what must be in ser.cfg ( if you've an example it could
>>>>>>>LB> help me a lot...)...
>>>>>>>LB> I wasn't able to find documentations about using Ser and Asterisk in
>>>>>>>LB> this configuration ( messages in the archives are not explicit
>>>>>>>LB> enough....) , so if you've a pointer or so....
>>>>>>>
>>>>>>>
>>>>>>>LB>    Help...
>>>>>>>
>>>>>>>LB> thx,
>>>>>>>LB> Laurent
>>>>>>>
>>>>>>>LB> _______________________________________________
>>>>>>>LB> Serusers mailing list
>>>>>>>LB> serusers at lists.iptel.org
>>>>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>    
>>>>>>>
>>>>>>>         
>>>>>>>
>>>>>>>              
>>>>>>>
>>>>>LB> _______________________________________________
>>>>>LB> Serusers mailing list
>>>>>LB> serusers at lists.iptel.org
>>>>>LB> http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>     
>>>>>
>>>>>          
>>>>>
>>>
>>> 
>>>
>>>      
>>>
>
>
>
>  
>